The accurate recognition of guitarist performance is challenging compared with other instruments because a guitar player typically plays several notes at once and uses both hands in different ways. In this paper, we propose a sensor-based guitar that consists of two groups of sensors. One sensor is used to recognize the fingering positions of the fretting hand, and the other is used to detect the guitar strings that are played by the picking hand. We design an embedded system for accurate sensing and propose a data analysis mechanism to precisely figure out the played pitch and the duration of notes using the sensed data. We realize our scheme as a high-quality prototype that detects guitarist performance with accuracy sufficient for the transcribing of a performance. We also present real application examples such as a rhythm game for interactive lessons and a music sharing feature with user created musical scores.

Providing seamless multimedia services to mobile users is one of the important requirements for the current Internet evolution. However, due to the effects of host mobility, it is difficult to achieve seamless QoS guarantees in mobile Internet. Though Next Step In Signaling (NSIS) protocol was recently proposed as an alternative of RSVP, its applicability has not been completely verified particularly in mobile environment. This paper enhances NSIS protocol based on advance resource reservation using layers 2 and 3 complementation for supporting host mobility. Our approach detects a crossover node (CRN) and reserves network resources proactively along the new path which will be used after handoff This significantly reduces the latency of signaling session re-establishment caused by handoff. Only a few modifications are required to the current NSIS protocol for our approach. The experimental results demonstrate that our approach can provide seamless QoS guarantees for multimedia services in mobile Internet.

This paper presents an automatic test generation and minimization method for testing interoperability of communication protocols such as the asynchronous transfer mode/broadband integrated services digital network (ATM/B-ISDN) signaling protocol and the transmission control protocol (TCP). The method is based on a technique of composing finite-state machines (FSMs). Traditionally, the generation of inputs utilizing FSM was mostly based on sequential test scripting languages and often done manually. Each input is processed on a one-at-a-time basis, which did not allow generating complex scenarios of simultaneous inputs on several interfaces of the system under test (SUT). In practice, there exists the possibility that additional inputs have to be sent to the SUT, while the previous input is still being processed, and/or multiple inputs need to be sent to different interfaces of the SUT at the same time. Our method generates interoperability test cases dealing with these more complex situations. Moreover, it minimizes the number of increased test cases due to the multiple stimuli without jeopardizing transition coverage. Experimental results have shown that compared with the conventional methods, our test generation method generates more interoperability test cases resulting in higher transition coverage (26% higher for TCP and 12% higher for ATM signaling protocol). Also, our test minimization method reduces the interoperability test cases by about 94% in the case of the ATM signaling protocol with the same transition coverage.

Supporting mobility in real-time applications such as Internet telephony is in demand as the mobile Internet becomes prevalent. A simple solution is to use Mobile JP in order to provide transparent handoffs to upper layer applications. However, due in part to the Mobile IP's limitations such as triangular routing delay and encapsulation overhead, application-layer solutions for mobile Internet telephony has been developed by extending Session Initiation Protocol (SIP). This work presents experimental results on Internet telephony based on Mobile IP and examines its feasibility by analyzing delay factors and increased network toads. We found that packet size in terms of number of frames per packet is important in optimizing the packet transmission delay and network load without hindering the interactivity in voice conversations. Our major contribution is to give a real experience on voice over Mobile IP and to raise a need of designing delay-aware and/or load-aware scheme for Internet telephony.